HowTo
OpenStage@Asterisk
Installation and Maintenance
Guide
Issue 1.0
Siemens Enterprise Communications GmbH & Co KG
Munich, 09/07/2010
Germany
Communication for the open minded
Siemens Enterprise Communications
Contents
Scope
Contents
2
3
Preparation
4
4
6
6
6
7
7
8
Supplying Power for the Phones
Connecting OpenStage Phones to the IP Network
802.1x
LLDP-MED Configuration Options
DHCP Options
Plug & Play – One Step Provisioning and Configuration
Single Phone Configuration (Local Menu, WBM)
Using OpenStage@Asterisk
Busy Lamp Function (BLF)
9
9
XML Applications
9
Send URL / Remote Server Control
Call Completion (CCBS/CCNR)
CTI for OpenStage - UACSTA
Changing the Caller Information – PAI Header
Multi Address Appearance (MAA)
Automatic Call PickUp – Using Alert Info Header
9
10
10
11
12
13
Logging and Tracing
LAN Port Mirroring
Tracing Capabilities within the Phone
Basic Troubleshooting
Local and Remote Tracing
QoS Data Collection
14
14
14
14
15
15
15
Remote Control - the HUSIM Phone Tester
Limitations
References
Abbreviations
17
17
17
Open Communications Principles and Best Practices
24/09/2010, page
3
Preparation
This chapter contains all information that is necessary to connect an OpenStage phone to an
Asterisk based communication system.
This includes the power supply options (PoE or external supply) for each OpenStage model
and its possible sidecar combinations. To enable a secure environment, 802.1x support for
OpenStage is specified.
For autoconfiguration, LLDP-MED and DHCP can be used. In addition to the standard DHCP
options, SEN proprietary enhancements allow for assigning the address of a provisioning
service at phone startup for easy Plug & Play installation. By means of the provisioning inter-
face (WPI), mass deployment scenarios and remote administration during the phone’s lifecy-
cle are supported.
To facilitate configuration of a single phone by the administrator or user, OpenStage phones
feature a web interface in addition to the phone’s local interface.
Supplying Power for the Phones
The OpenStage phone family can be powered by
•
•
External power supply
Power over Ethernet (PoE)
Feature
OpenStage
15
OpenStage
20E
OpenStage
40
OpenStage
60
OpenStage
80
OpenStage
20
OpenStage
40 G
OpenStage
60 G
OpenStage
80 G
OpenStage
20 G
Legacy optiPoint
external Power Sup-
ply (EU-, US- or UK-
plug)
Yes (op-
tional)
Yes (op-
tional)
Yes (op-
tional)
Yes (op-
tional)
Yes (op-
tional)
New OpenStage
Power Supply (EU-,
US- or UK-plug)
Yes (op-
tional)
Yes (op-
tional)
Yes (op-
tional)
Yes (op-
tional)
Yes (op-
tional)
In line with the further development of our portfolio we are now able to offer
smaller and lighter power supplies for all OpenStage and optiPoint 410/ 420/
500 variants.
They feature a higher degree of efficiency leading to 14 - 19% lower power
consumption, depending on the connected devices.
Important: The new OpenStage power unit is a device of product category.
The protection concept of the power installation needs to achieve the re-
quirements of earth conductor (e.g. Schukostecker Typ F, country specific).
Power over Ethernet:
IEEE 802.3af
Yes
Yes
Yes
Yes
Yes
Power over Ethernet,
Cisco proprietary
mode
No
No
No
No
No
802.3af Power class
OS 15: 1
OS 20E : 1
OS 20 : 1
OS 20 G: 2
OS 40: 2
OS 40 G: 3
OS 60 : 3
OS 60 G: 3
OS 80 : 3
OS 80 G: 3
Open Communications Principles and Best Practices
24/09/2010, page
4
Energy saving mode
To reduce the energy consumption to a minimum, OpenStage phones offer an energy saving
mode. The display backlight (phone and Key Module, if attached) is switched off after a con-
figurable timeout.
With OpenStage 40, the main display and key module backlight will be switched off after 90
seconds of inactivity (firmware version V2R0 onwards). Readability even without backlight is
ensured by the transflective display.
With OpenStage 60 and 80, the timer is configurable by the administrator (Local Functions
> Energy saving); the timeout ranges between 2 and 8 hours.
Power consumption [W] - Fast Ethernet variants
Energy saving
mode
Idle state
During call
(handset)
Ringing (max.
vol.)
OpenStage 15
OpenStage 15 with
1 OpenStage Key
Module 15
-
2,0
2,2
2,9
3,1
2,0
2,9
3,1
2,3
3,2
3,4
3,9
4,1
4,1
4,3
1
2,2
2,5
(9 LED’s on)
OpenStage 20/20E
OpenStage 40
OpenStage 40 with
1 OpenStage Key
Module 40
OpenStage 40 with
OpenStage Key
Module 40
-
-
1,8
1,9
2,6
2,8
1,8
2,4
2,6
3,5
2,0
2,7
2,8
3,6
2,9
4,0
3,4
4,4
1
2
1
2,1
2,3
3,0
3,2
3,1
3,8
4,2
4,9
3,4
4,1
4,4
5,2
4,6
5,2
5,1
5,8
OpenStage 40 with
1 OpenStage Key
Module 15
2,1
3,0
2,6
3,7
2,9
3,8
4,2
4,6
(9 LED’s on)
OpenStage 60
OpenStage 60 with
1 OpenStage Key
Module 60
-
2,4
2,6
3,3
3,5
5,6
6,2
6,9
7,6
5,8
6,5
7,1
7,9
6,8
7,6
7,9
8,5
1
OpenStage 60 with
2 OpenStage Key
Module 60
OpenStage 80
OpenStage 80 with
1 OpenStage Key
Module 80
2
-
2,8
2,3
2,5
3,7
3,2
3,4
6,9
6,3
7,0
8,3
7,7
8,5
7,1
6,5
7,2
8,6
7,9
8,7
8,3
7,6
8,4
9,3
8,6
9,4
1
OpenStage 80 with
2 OpenStage Key
Module 80
2
2,6
3,6
7,6
9,2
7,9
9,5
9,1
10,2
Open Communications Principles and Best Practices
24/09/2010, page
5
Power consumption [W] - Gigabit Ethernet variants
Energy saving
mode
Idle state
During call
(handset)
Ringing (max.
vol.)
OpenStage 20 G
OpenStage 40 G
-
-
1
2
-
-
-
-
-
1
-
3,6
3,8
4,0
4,1
4,6
3,8
4,0
4,7
4,9
5,1
5,3
5,8
4,9
5,1
3,6
4,7
5,6
6,4
7,3
6,6
8,7
9,3
4,0
5,0
5,9
6,8
7,7
7,0
9,0
10,0
4,5
5,2
6,2
6,9
6,4
7,8
9,3
10,1
*)
8,4
9,4
10,3
*)
5,3
6,3
7,4
7,9
7,4
9,4
10,3
4,4
5,1
5,9
5,3
7,0
7,7
4,7
5,5
6,3
5,6
7,4
8,3
OpenStage 60 G
Please see note *)
OpenStage 80 G
Please see note *)
-
1
-
2
-
4,2 *)
5,3
8,4 *)
10,0
9,0 *)
10,7
11,0
-
1
-
-
3,6
4,4
4,9
5,6
7,4
8,1
9,2
9,8
7,9
8,4
9,6
10,1
10,0
10,6
2
-
4,5 *)
5,7
8,8 *)
10,6
9,2 *)
10,9
11,5
*) These values are still within the 802.3af PD class 3, which allows up to 12,95 W, but they
are averaged, not maximum values. As soon as the USB interface is used, PD class 3 is ex-
ceeded. Therefore, an external power supply has to be used for an OpenStage 60G/80G with
2 Key Modules.
Connecting OpenStage Phones to the IP Network
802.1x
OpenStage phones support 802.1x EAP-TLS. Certificates for authentication can be
downloaded via the WPI.
LLDP-MED Configuration Options
OpenStage SIP phones support the layer 2 protocol LLDP-MED (Link Layer Discovery Protocol-
Media Endpoint Discovery). It is used for simplification of auto-configuration and network
management. The auto-configurable parameters of OpenStage phones are mainly the VLAN
ID, power consumption (power class) and quality of service (QoS) parameters.
LLDP-MED is able to replace various other established mechanisms like DHCP options, man-
ual configuration, or proprietary solutions like Cisco CDP. Example parameters are LAN speed
and duplex discovery, network policy discovery (VLAN and QoS capabilities) or extended
power via MDI discovery (PoE).
When an OpenStage phone is connected to a switch with LLDP-MED capabilities, the phone
is able to
a) advertise and receive a VLAN ID,
Open Communications Principles and Best Practices
24/09/2010, page
6
b) advertise and receive QoS parameters,
c) advertise the power requirements to the LAN access switch by means of an "Ex-
tended Power via MDI” TLV.
Note: LLDP-MED should only be used with LLDP enabled network access switches. Older
network access switches that don't adhere to the 802.1D-1998 MAC bridging specification
might appear to be propagating the LLDP multicasts through the subnet. In this case, LLDP-
MED should be deactivated on the phone. For further information, please refer to the Open-
Stage Asterisk Admin Guide [3].
DHCP Options
The following parameters can be obtained by DHCP:
Basic Configuration
•
•
IP Address
Subnet Mask (option 1)
Optional Configuration
•
•
•
•
•
•
•
•
Default route (option 3)
Static IP routing (option 33)
SNTP server (option 42)
Timezone offset (option 2)
Primary/secondary DNS server (option 6)
DNS domain name (option 15)
SIP Addresses / SIP Server & Registrar (SIP Server option 120)
Vendor unique (option 43)
The vendor specific option (code 43), or alternatively, a vendor class, is used to provide the
phone with the location of an optional configuration/ provisioning service. By this means,
full Plug & Play is possible (see the following section). For further information, including an
example configuration for dhcp, please refer to the OpenStage Asterisk Admin Guide [3].
Plug & Play – One Step Provisioning and Configuration
Provisioning via the WPI (WorkPoint Interface)
The WPI allows for controlling and provisioning the phone by a remote service using XML
messages which are transmitted over HTTPS. Unlike many other VoIP phones, which are
limited to prefabricated configuration files for download at startup time, OpenStage phones
exchange information with the provisioning service continuously. When local changes have
been executed on the phone, it will inform the provisioning service automatically.
Any kind of administration task is supported, for instance, updating the firmware on a selec-
tion of phones, or setting SIP codes for server-based telephony features.
For further information, please refer to the WPI Developer’s Guide [8].
Plug & Play / Autoprovisioning
A fully automated mass rollout of OpenStage phones can be realized by combining a DHCP
server and the provisioning service. On startup, the phone receives the IP address of the
provisioning server from the DHCP server. After that, it contacts the provisioning service. The
provisioning service may then request all settings from the phone in order to decide which
Open Communications Principles and Best Practices
24/09/2010, page
7
parameters must be set or updated. When all these parameters have been sent to the phone,
it is ready for operation.
For further information, please refer to the WPI Developer’s Guide [8].
If a Firewall or NAT get in the Way
In case the phones and the provisioning service reside in different networks or subnets that
are separated by a firewall and/or NAT, it may be impossible for the provisioning service to
contact the phones.
To enable a solution for this problem, the phone can be configured to periodically poll
the provisioning service, or a special proxy, for new messages. Thus, provisioning service
driven interactions are possible even when the provisioning service is located behind a fire-
wall, or in a DMZ. For further information, please refer to the WPI Developer’s Guide [8],
Section 1.4.4.3, "Polling Request To Bridge A Firewall" and Section 3.1.2.2, "Provisioning
Service Located Behind A Firewall".
Single Phone Configuration (Local Menu, WBM)
Generally, it is recommended to administrate and configure an OpenStage phone installation
remotely using a provisioning service via the phone’s WPI (WorkPoint Interface). However,
there are two further configuration interfaces; these can be used to administrate individual
phones:
•
•
Local menu: The user interface of the device itself.
Web Based Management (WBM): The phone’s web interface.
Note: The default password for OpenStage administration is <123456>.
Local Menu
The phone’s application key ( ) is used to access the user and administration menu at the
phone.
Web Based Management (WBM)
The phone’s web interface can be accessed by any common web browser, like Firefox or
Internet Explorer. As HTTPS is used, the URL must be entered as follows:
https://<phone-ip-address>
If the browser displays a certificate notification, accept it.
Alternatively, the DNS name of the phone can be entered, if it has been configured and DNS
is available in the network.
Open Communications Principles and Best Practices
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8
Using OpenStage@Asterisk
This chapter contains some tips and tricks for operating OpenStage phones with Asterisk. It
is not intended to be exhaustive, but provides some major topics and solutions from differ-
ent successful customer projects.
Busy Lamp Function (BLF)
The “Busy Lamp Field” feature (available for OpenStage 15/40/60/80) allows users to monitor
the dialog state of another phone via the LED associated to an FPK. Please note that, some-
times, the term 'Direct Station Selection' is used for the same functionality.
Depending on which function is configured for the FPK, the user can also pick up a call for
another user, which is of much use in working teams. The feature is described in detail here:
isk_Feature_Busy_Lamp_Field_%28BLF%29#For_Users
XML Applications
OpenStage 60/80 phones feature a graphical user interface and an XML application platform,
which allows for developing custom applications based on HTTP/HTTPS. The phone acts as a
front-end for a server-side program. The interaction can be initiated by the phone or by the
server-side program. Besides displaying, modifying, and submitting data, XML applications
have the capability of starting calls on the phone. Possible uses for OpenStage XML Applica-
tions might be
•
Integration with groupware (e.g. Microsoft Exchange Server) or Unified Messaging
systems (e.g. Siemens Enterprise Communications OpenScape)
phonebooks with access to address databases
call recording
presence applications
collecting information provided by web services (e.g. news, weather, traffic, stocks)
attendance clock
•
•
•
•
•
For detailed information, please refer to:
enterprise.com/developerportal/Resource%20Center/OpenStage%20XML.aspx
Send URL / Remote Server Control
The FPKs and FFKs (firmware V2R1) can be utilized for communication with a server-side
program.
To enable feedback from the server, the LED associated with the key can be controlled re-
motely. This can be done via SIP notify messages, or, with firmware version V2R2 onwards,
via a combination of HTTP push requests and XML documents. For detailed information,
please see the
OpenStage XML Applications Developer's Guide [9],
and
enterprise.com/developerportal/Resource%20Center/OpenStage%20XML.aspx
Open Communications Principles and Best Practices
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9
Call Completion (CCBS/CCNR)
Call completion is a telephony feature which takes action on a failure to complete a call. It
allows for notifying the calling user when the called user is available again.
The OpenStage callback feature covers two conditions for call completion:
CCBS (Call Completion Busy Subscriber) : The called party is busy.
CCNR (Call Completion No Reply) : The called party does not respond.
Call Completion features can be implemented on a PBX, a dedicated server (e.g. a voicemail
Server) or directly on the client device (e.g. messaging applications).
There are several commercial companies which provide call completion features, as well as
IETF documents specifying call completion features for open standards, such as SIP.
The RFC 5359 gives a best practice example for call completion. However, the SEN OSCAR
group has evaluated this RFC with the result that it is not useful for a B2BUA architecture.
ietf-bliss-call-completion-06.txt) on how call completion can be implemented. However, no
RFC is available for this topic yet.
Although no standard has been released until now, OpenStage phones already support
server based call completion.
An implementation according to the IETF BLISS draft is planned, but currently not available.
The OpenStage call completion implementation is purely stimulus based and can be found
at:
CTI for OpenStage - UACSTA
There are several use cases where remote control of a VoIP phone is required. Among these
are server based features like ‘Call Forwarding’ or ‘Do not Disturb’, or agent desktop applica-
tions requiring a seamless desktop integration of the phone.
There is one ECMA standard in place which covers all those requirements: uaCSTA.
By means of the uaCSTA interface, the OpenStage SIP user agent can use call and device
control services at the SIP Server and vice versa. A complete set of CSTA services are defined
in ECMA-269 [6], which should be referenced for additional information.
The following subset of CSTA services and events are supported by OpenStage:
Î Services on the SIP Server:
¾
¾
¾
¾
Set Forwarding
Get Forwarding
Set Do Not Disturb
Get Do not Disturb
Î Events Generated by the SIP Server:
Forwarding Event
¾
Open Communications Principles and Best Practices
24/09/2010, page 10
¾
¾
Do Not Disturb Event
Diverted Event
Î Services on the OpenStage device:
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
Make Call
Answer Call
Hold Call
Retrieve Call
Clear Connection
Consultation Call
Generate Digits
Get Volume
Set Volume
Get Mute
Set Mute
Î Events Generated by OpenStage:
OpenStage does not generate CSTA Events.
¾
With these services a SIP server can easily control basic OpenStage functions. For futher in-
formation please have a look at:
enterprise.com/images/e/e7/white_paper_uaCSTA_Public_version_2010803.pdf
Changing the Caller Information – PAI Header
SIP is a great protocol for call processing. However, in some use cases, additional and up-to-
date information about a caller might prove to be very useful. Among the possibilities are:
¾
¾
¾
¾
Add caller information from an external database
Update caller information during call transfer
Add hunt group information for incoming calls
Enhance Executive/Assistant features with additional information
OpenStage supports RFC 3325 [7]. A SIP server can change the OpenStage display informa-
tion using SIP INVITE or UPDATE requests or any SIP response code.
Especially the P-Asserted-Identity Header can be used to carry additional information to the
phone user.
Open Communications Principles and Best Practices
24/09/2010, page 11
Multi Address Appearance (MAA)
A telephone is normally associated with a directory number, or, generally, with a SIP AoR.
This number is used for placing calls to the associated telephone and for displaying the tele-
phone's (user's) identity when placing calls to another party. The number is also used when
more than one call appearance is supported due to additional features like call waiting.
The term ‘keyset’ denotes a telephone which is associated with more than one number. This
allows a given telephone to act on behalf of other phone numbers, resp. users. Just like with
traditional telephony systems, people sometimes refer to lines instead of numbers, hence
keyset phones are also referred to as ‘multiline phones’. The main line, i.e. the line/directory
number associated with a given physical telephone, is called primary line. All other lines,
which can also be handled on other phones, are denoted as secondary lines. Please note that
call log and MWI (Message Waiting Indication) are supported only for the primary line, not
for secondary lines.
The programmable feature keys are used for handling the lines and their respective call ap-
pearances, supported by the associated LEDs reflecting the line/call status. The number of
lines that can be configured is depending on the phone model. For OpenStage 60/80, up to
30 lines can be configured. OpenStage 15/20/40 are limited to 18 lines.
This feature can be used for different use cases, for example:
¾
¾
¾
Address multiple users at one phone
Enhanced call hold scenarios
Allow more than two incoming calls at one phone
For some use cases, however, this feature can not be used, for example:
¾
¾
MLA. If the line is configured at more than one phone, incoming calls are sent to the
last registered device
Line status observation. If the same line is configured at more than one phone, the
line status is not presented at these phones.
MAA is automatically activated if line keys are configured at the phone. Line key administra-
tion is done by the administrator; the user has no influence on these settings. The phone
operates as an MAA phone ‘out of the box’. Depending on the administrator settings, the
phone will react slightly different in basic user interactions.
Even if only one line key is configured, the phone changes into line presentation mode. The
line presentation mode helps the user to keep track of the different line statuses.
Open Communications Principles and Best Practices
24/09/2010, page 12
Example: OpenStage operates in MAA mode.
Further information can be found at:
Automatic Call Answering Using Alert-Info Header
Besides using uaCSTA, the phone can be set to automatically answer a call by adding an alert
info header to the call. Thus, the SIP server is enabled to control whether a call is to be an-
swered immediately and without user interaction. However, the user has the possibility to
allow or disallow this feature by setting User Pages > Configuration > Incoming calls > CTI
calls > Auto answer.
The Alert-Info header must be set as follows:
Open Communications Principles and Best Practices
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Logging and Tracing
OpenStage phones are perfect. But if something should go wrong anyhow, the customer
service needs tools to focus on the problem. Service effort is needed, but should be mini-
mized. Therefore, OpenStage phones provide plenty of tools and options to find the cause of
a problem quickly, even if it is not located at the phone.
LAN Port Mirroring
Every OpenStage phone has a built-in Ethernet switch. One of the ports is used to connect
the phone to the local network. The other port is intended for connecting a PC, thus allowing
network connectivity for both devices with only one wire from the desk.
In addition, the PC port enables network monitoring which might be useful for development
and error tracing. For this purpose, the PC port must be configured as a mirror for the LAN
port by setting PC port mode to “mirror” (see [3]). If configured this way, the complete traf-
fic of the LAN port will be passed through to the PC port, just like with a simple network hub.
Now, a network tracing tool on the PC can trace all IP traffic, like SIP over UDP, or XML over
HTTP, for instance.
Tracing Capabilities within the Phone
OpenStage phones provide strong support for system integration, testing, and troubleshoot-
ing. Besides the Administration Guide [3], the tracing capabilities of the phone are described
in [5].
Basic Troubleshooting
For tracking network issues, the phone can execute ping and traceroute tests; these can be
controlled and viewed online using the WBM.
For elementary troubleshooting, the phone provides an overview about basic issues in the
user menu. The admin can ask the user to read that basic information to get a first hint
about the possible causes of an issue:
Problem Description
Error code
Network Problem No network connection
Not Initialised Waiting for data
Unable to use LAN 802.1x error
LI1
I1
LX1
Unable to use LAN Physical connection missing LP1
Unable to Register Server timeout
Unable to Register Server failed
Unable to Register Authentication failed
Unable to Register No number configured
Unable to Register No server configured
Unable to Register No registrar configured
RT2
RF2
RA2
RN2
RS2
RG2
Unable to Register No DNS domain configured RD2
Unable to Register Rejected by server
Unable to Register No phone IP address set
Survivability Backup route active
RR2
RI2
B8
Survivability Backup not configured
Survivability Backup timeout
Survivability Backup authentication failed
RS8
RT8
RA8
Open Communications Principles and Best Practices
24/09/2010, page 14
Local and Remote Tracing
The phone is able to write internal trace files, and to send the trace data to a remote syslog
server. The tracing can be configured in a differentiated way by setting discrete trace levels
for each service.
Please note that it is not recommended to enable all traces to the deepest level. The gener-
ated trace file will exhaust the phone’s memory shortly, and the overall functionality will
slow down.
QoS Data Collection
OpenStage phones generate QoS reports using a HiPath specific format, QDC (QoS Data Col-
lection).
The reports created for the last 6 sessions, i. e. conversations, can be viewed on the WBM or
are reported to the QCU (QoS data Collection Unit).
SEN provides a server application to collect the data. The collected data is sent via SNMP. If
an SNMP server is available, the QDC MIBS can be downloaded from our software supply
server (SWS).
Meanwhile, third party solutions are available which can also deal with the OpenStage QDC
data.
Remote Control - the HUSIM Phone Tester
Sometimes everything goes wrong: tracing is of no help, issues are sporadic and the cus-
tomer’s problem cannot be understood. It seems that the last resort is a visit to the customer
to get the needed information.
In such situations, the HUSIM phone simulator can produce relief. It enables the service staff
to access a defined group of phones remotely. The tool works similar to well known remote
control tools for PCs like VNC.
For each phone, a PC application window shows the current status. Every OpenStage phone
model is represented with its complete key layout and display content. The remote visitor
can see all user interactions on the phone. Moreover, he can access the phone keys actively
and in this way operate the phone by remote control. Please note that, for privacy protec-
tion, the user is always informed about the remote interaction.
To get the phone tester up and running, a special dongle key must be uploaded to the
phone. The dongle key and the HUSIM software can be downloaded without additional
charge from SWS/SEBA. The key can be distributed to the phone using the SEN DLS (Deploy-
ment Service) or the phone’s WPI (WorkPoint Interface).
Open Communications Principles and Best Practices
24/09/2010, page 15
Example Screen: OpenStage 80 represented in the HUSIM Phone Tester
Open Communications Principles and Best Practices
24/09/2010, page 16
Limitations
Not known yet :-)
References
[1] TIA-811-A: Performance and Interoperability Requirements for Voice-over-IP (VoIP) Fea-
ture Telephones (http://www.tiaonline.org/standards/technology/voip/documents/TIA-
811-A-final-for-global.pdf)
[2] Session Initiation Protocol (SIP)-Specific Event Notification: (RFC 3261)
enterprise.com/images/e/e1/Administration_Manual_OpenStage_Asterisk.pdf)
enter-
prise.com/images/c/c7/OpenStage_Provisioning_Interface_Developer%27s_Guide.pdf)
enterprise.com/images/1/1b/Service_Info_How_to_trace_OST_SIP.pdf
[6] Services for Computer Supported Telecommunications Applications (ECMA-269).
[7] Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity
enter-
prise.com/images/c/c7/OpenStage_Provisioning_Interface_Developer%27s_Guide.pdf
enterprise.com/gdmsproxy/retrieve?id=40776497
Abbreviations
AoR
BLA
Address of Record
Bridged Line Appearance
CCBS Call Completion Busy Subscriber
CCNR Call Completion No Reply
CTi
FFK
FPK
Computer Telephony Integration
Free Function Key
Free Programmable Key
HTTP Hypertext Transfer Protocol Overview
HTTPS HyperText Transfer Protocol Secure
MAA Multiple Address Appearance
MLA
MSA
MWI
SCA
SIP
Multiple Line Appearance
Multiple Station Appearance
Message Waiting Indication
Shared Call Appearance
Session Initiation Protocol
User Agent
UA
uaCSTA User Agent Computer Supported Telephony Application
Open Communications Principles and Best Practices
24/09/2010, page 17
About Siemens Enterprise Communications Group (SEN Group)
©Siemens Enterprise
Communications GmbH & Co. KG
The SEN Group is a premier provider of enterprise communications solutions. More than 14,000 employees in 80 coun-
tries carry on the tradition of voice and data excellence started more than 160 years ago with Werner von Siemens and
the invention of the pointer telegraph. Today the company leads the market with its "Open Communications" approach
that enables teams working within any IT infrastructure to improve productivity through a unified collaboration experi-
ence. SEN Group is a joint venture between the private equity firm, The Gores Group, and Siemens AG and incorporates
Siemens Enterprise Communications, Enterasys Networks, SER Solutions, Cycos and iSEC.
Siemens Enterprise
Communications GmbH & Co. KG
is a Trademark Licensee of Siemens AG
Status 03/2009
The information provided in this brochure contains merely
general descriptions or characteristics of performance
which in case of actual use do not always apply as described
or which may change as a result of further development
of the products. An obligation to provide the respective
characteristics shall only exist if expressly agreed in the
terms of contract. Availability and technical specifica-
tions are subject to change without notice. OpenScape,
OpenStage and HiPath are registered trademarks of
Siemens Enterprise Communications GmbH & Co. KG.
All other company, brand, product and service names are
trademarks or registered trademarks of their respective
Communication for the open minded
Siemens Enterprise Communications
holders.
Printed in Germany.
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